With pjsip (asterisk 13.14.1) I see the problem that an anonymous from header gets user=phone appendend to the URI if user_eq_phone=yes is specified:On the incoming leg:From: anonymous ;tag=Q5zBj7BMnvI6Fe6O2866fox3ZHmn-smt Get transformed to From: Anonym..

Having enabled a strict DMARC setup I noticed everytime I send a message here I get all these reports of messages which fail DMARC. Since I dont want people to miss my wise thoughts maybe the maintainers of this list could look into DKIM signing ..

Im trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures.For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesntS..